Difference between revisions of "H323 analysis"

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== H323 Protocol stack ==
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([http://www.en.voipforo.com/H323/H323_protocolos_stack.php source])
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The most known protocols used in H.323 are:
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* RTP/RTCP (Real-Time Transport Protocol / Real-Time Transport Control Protocol): Internet-standard protocol for the transport of real-time data, including audio and video. RTP is used in virtually all voice-over-IP architectures, for videoconferencing, media-on-demand, and other applications. A thin protocol, it supports content identification, timing reconstruction, and detection of lost packets.
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* RAS (Registration, Admission and Status): A protocol for Registration, Admission and Status. In an H.323 audio or video system, the RAS is a control channel over which H.225.0 signaling messages are sent.
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* H225.0: Protocol used to describe call signaling, the media (audio and video), the stream packetization, media stream synchronization and control message formats
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* H.245: Control protocol for multimedia communication, describes the messages and procedures used for opening and closing logical channels for audio, video and data, capability exchange, control and indications
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It manages the following functionalities:
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# Interchange of capacities: Terminals define the codecs they have and they send it to the other end point.
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# Opening and closing logical channels: H.323 audio and video channels are point to point and unidirectional. Therefore they will have to create at least two of these channels. This is responsibility of H.245.
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# Flow Control when there is a problem.
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# A lot of different small functions.
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* Q.931: A protocol for Call Signaling, consisting of Setup, Teardown and Disengage. Q.931 is included in the H.225.0 Recommendation
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* RSVP (Resource ReSerVation Protocol): Protocol for reserving network resources to provide guaranteed application QoS (Quality of Service)
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* T.120: Standard for data conferencing and conference control for interactive multimedia communication - multipoint & point-to-point.
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The following codecs are recommended by the H.323 standard:
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* G.711: ITU-TSS recommendation "Pulse code modulation (PCM) of voice frequencies". This audio standard is mandatory for all video conferencing systems. It requires a data rate of 56 or 64 kbit/s.
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* H.261y H.263: Video codecs of H.323 standard. However, other ones can be used.
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== Client possibilities ==
 
== Client possibilities ==
 
For some possible H323 (and SIP) VoIP clients you can check [http://forskningsnett.uninett.no/voip/voipclients.html this list]
 
For some possible H323 (and SIP) VoIP clients you can check [http://forskningsnett.uninett.no/voip/voipclients.html this list]

Revision as of 15:55, 19 February 2007

H323 Protocol stack

(source)

The most known protocols used in H.323 are:

  • RTP/RTCP (Real-Time Transport Protocol / Real-Time Transport Control Protocol): Internet-standard protocol for the transport of real-time data, including audio and video. RTP is used in virtually all voice-over-IP architectures, for videoconferencing, media-on-demand, and other applications. A thin protocol, it supports content identification, timing reconstruction, and detection of lost packets.
  • RAS (Registration, Admission and Status): A protocol for Registration, Admission and Status. In an H.323 audio or video system, the RAS is a control channel over which H.225.0 signaling messages are sent.
  • H225.0: Protocol used to describe call signaling, the media (audio and video), the stream packetization, media stream synchronization and control message formats
  • H.245: Control protocol for multimedia communication, describes the messages and procedures used for opening and closing logical channels for audio, video and data, capability exchange, control and indications

It manages the following functionalities:

  1. Interchange of capacities: Terminals define the codecs they have and they send it to the other end point.
  2. Opening and closing logical channels: H.323 audio and video channels are point to point and unidirectional. Therefore they will have to create at least two of these channels. This is responsibility of H.245.
  3. Flow Control when there is a problem.
  4. A lot of different small functions.
  • Q.931: A protocol for Call Signaling, consisting of Setup, Teardown and Disengage. Q.931 is included in the H.225.0 Recommendation
  • RSVP (Resource ReSerVation Protocol): Protocol for reserving network resources to provide guaranteed application QoS (Quality of Service)
  • T.120: Standard for data conferencing and conference control for interactive multimedia communication - multipoint & point-to-point.

The following codecs are recommended by the H.323 standard:

  • G.711: ITU-TSS recommendation "Pulse code modulation (PCM) of voice frequencies". This audio standard is mandatory for all video conferencing systems. It requires a data rate of 56 or 64 kbit/s.
  • H.261y H.263: Video codecs of H.323 standard. However, other ones can be used.


Client possibilities

For some possible H323 (and SIP) VoIP clients you can check this list The only two that support both H323 and linux are:

Another option is the commandline ohphone. No graphical interface.

Ekiga

If it were to work the interface is supposed to look like this To be downloaded from the ekiga page. Unfortunately the packages OPAL and PWLIB libraries have to be installed first. This is a tedious and time comsuming process (lots of compiling needed). When we were finally done with this we tried to configure ekiga but we got the error message: Package gnome-doc-utils was not found in the pkg-config search path. We gave up since no system manager was available at this moment and decided to go on exploring the SJPhone option first.

Gnomemeeting

Today (1-31) the system manager installed all the necessary packages. But there is a problem for Ekiga when locating the audio-plugins; failing this stops the program execution. Compilation from the sources has been tried as well; unfortunately, after compiling and installing several libraries, the process got stuck on pwlib, where Ekiga requires version 1.10.3 but Debian provides only 1.10.0. Another installed H323 client is the predecessor of Ekiga, Gnomemeeting. This is already installed on the system and seems to work. So far we don't know if this is working; the application starts and everything seems ok, sound support as well. So far no H323 extension is plugged in into TrixBox; this is the next step to support H323 communication.

SJPhone

Very straightforward download from SJ Labs (VoIP software). You get the following interface.

Shermafdruksjphone.png

Note: At least the Windows version of SJ-phone is still using SIP, even when this is disabled in the configuration. When using 2 instances of SJphone, to call eachother, the connection is true H323.

H323 and TrixBox

We tried to look up some information on H323 and TrixBox. From the information on the web it seems that this is a tricky topic. Many questions, few answers. People seem to have trouble with it. I haven't been through all the correspondence on the TrixBox forum but it seems that most people have a hard time even setting up a connection (lots of configuring) and when they do succeed the connection dies after 15-30 seconds. We will look into this a bit more tomorrow.


H323-related links